VOIP Quick Guide

VO IP A rch ite c tu re
Session Initiation Protocol (SIP) SIP
Terminal
Network Architecture H.323
Terminal

SIP H.323
Phone LAN Phone
LAN H.323
Router Network Architecture
SIP Router
SIP Server H.323
Terminal Gatekeeper H.323
Internet Terminal

SIP Router
LAN Server H.323
LAN Phone

Router
GCP Router
Phone
Media MSU
Gateway H.323
SoftSwitch Router
Gatekeeper
Signaling WAN
Gateway

SS7 Signaling
SoftSwitch
Router GCP Signaling
Gateway
Gateway SIP Signaling
GCP Gateway
Phone PSTN
Media H.323 Signaling
Gateway H.324
Gateway Multimedia RTP Traffic
over POTS
GCP PSTN Traffic
Class 5
Terminal Switch
POTS H.320
Gateway Control Protocol (GCP) Phone Multimedia
Network Architecture ISDN over ISDN
Phone

VO IP Pro to c o ls
Applications Media Control Signaling

AAA Audio Codecs Video Codecs GCP SIP H.323 Other

Authentication Authorization Accounting
IPDR OSP G.711 G.728 H.261 H.248 Cisco SCCP
SIP H.245
Internet Protocol Open Settlement MEGACO Skinny Client
Session Call Control
Detail Record G.722 G.729 H.263 Media Gateway Control
Protocol Initiation
Control Protocol Protoc ol
RADIUS G.723.1 G.726 H.264 MPEG-4 H.225.0
Remote Authenticaion Dial-In User Service Q.931
MGCP
SDP Call Setup NCS
RTSP RTCP Media Gateway
Session Network-Based
Services Real Time Real Time Control Protocol
Description H.225.0 Call Signaling
Streaming Control
H.450 Protocol RAS
Real-time Protocol Protocol SGCP
Supplement Registration
Conferencing Simple Gateway
Services SCTP RTP Admission Skype
T.120 Control Protocol
(e.g.,Call Waiting) Stream Control Real Time Status Signaling
SAP
Transmission Transport Session
FoIP MoIP Protocol Protocol IPDC
Announcement H.235
Facsimile over IP Modem over IP IP Device
Protocol Security
T.38 V.150 cRTP Control
Compressed Real Time Protocol

TCP/UDP UDP/RUDP TCP/UDP

IP
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 VOIP Technology Comparison
H.323 SIP MGCP/H.248/Megaco

Standards body ITU-T IETF MGCP/Megaco by IETF; H.248 by ITU-T

Architecture Distributed Distributed, Peer-to-Peer Centralized

Call control Gatekeeper Proxy/Redirect Server Call agent/Media Control Gateway / Softswitch

Endpoints Gateway, terminal User agent Media Gateway, dump terminal

MGCP - UDP
Signaling transport TCP/UDP TCP/UDP
H.248/Megaco-TCP/UDP

Multimedia Yes Yes Yes

RTP-Real Time RTP-Real Time RUDP - Reliable User Datagram Protocol;

Media transport
Transport Protocol Transport Protocol RTP - Real Time Transport Protocol

DTMF-relay transport RTP RTP RTP

Fax-relay transport T.38 T.38 T.38

Supplemental services By endpoints or call control By endpoints or call control By call agent

H.323 - A Distributed VOIP Network

H.323 Network Elements
Terminals: a network endpoint which may provide audio, data and video, communications with another H.323 terminal.
Gateways: a network function that provides access to terminals on a circuit switched network (such as the PSTN) or another H.323 network.
Gatekeepers:a network function that provides address translation, access control, bandwidth management, and other management operations.
Multipoint Control Units:a network function that allows three or more terminals to participate in a multipoint conference.

Example H.323 Call Flow ITU-T H.323 Standards

Standard# Description

An umbrella recommendation of ITU-T that

H.323 defines the protocols to provide audio-visual
H.323 gatkeeper
H.323 gateway communication sessions on any packet network.
H.225 TCP connection For call signaling, the media (audio and
H.225.0 video), the stream packetization, media
stream synchronization and control message
SETUP formats.

H.225 H.235 For secur ity in H.323 networ k.

CONNECT (H.245 address) signaling

For dual stream use in

H.239 videoconferencing.

A control protocol for multimedia

H.245 communication.
H.245 TCP connection

Capabilities exchange H.246 H.323/PSTN Interworking.

RTCP address Directory Services Architecture

H.350
for Multimedia Conferencing.

CONNECT (H.245 address) H.245 An architecture for end-to-end

signaling H.360 QoS control and signaling.
RTCP addresses
For supplementary services such as call
H.450 waiting, call forwarding, etc.
RTCP&RTP addresses

H.460.x Supplements in H.323.

RTP stream Protocol for mobility management and

H.501 intra/inter-domain communication
RTP stream in multimedia systems.

H.323 gateway IP phone H.510 Mobility for H.323 multimedia systems.

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SIP: Session Initiation Protocol A Peer-to-Peer VOIP Network
SIP Network Elements
User Agents: A software program installed in a user’s terminal or an IP phone to initiate and terminate phone calls, plus data and video communications.
There are two logic parts in the user agents: User Agent Server (UAS) and User Agent Client (UAC). UAC sends requests and receives responses. UAS
receives requests and sends responses.
Proxy Server: Performs routing of a session invitations according to invitee's profile. There are two basic types of SIP proxy servers--stateless and stateful.
Stateless servers are simple message forwarders. Stateful proxies, upon reception of a request, create a state and keep the state until the transaction finishes.
Redirect Server: Receives a request and sends back a reply containing a list of the current location of a particular user, by looking up the intended recipient of
the request in the location database created by a registrar.
Registrar Server: A registrar is a server that accepts REGISTER requests and places the information it receives in those requests into the location service for
the domain in handles.

Example SIP Call Flow IETF SIP Standards

RFC# Description

2974 Session Announcement Protocol (SAP)

2976 The SIP INFO Method

SIP Phone A SIP PROXY SIP Phone B
3261 SIP: Session Initiation Protocol
(updated by RFC 3853, RFC 4320)
SIP /SD P INV ITE

g SIP/ SDP INVI TE

St atu s:1 00 Tr yin
3262 Reliability of Provisional Responses in SIP

ss ion Pr og re ss 3263 SIP: Locating SIP Servers

St at us :1 83 Se
on Pro gre ss
Sta tus :18 3 Se ssi
3265 SIP-Specific Event Notification

St at us :2 00 OK
St at us :2 00 OK
3311 SIP UPDATE Method

Private SIP Extensions for Media

3313
Authorization
SIP AC K
SIP Extension for Registering
SIP AC K 3327
Non-Adjacent Contacts
Security Mechanism Agreement
3329
for SIP Sessions
RTP/RTCP stream
3420 Internet Media Type message/sipfrag

3428 SIP Extension for Instant Messaging

SIP :BY E
SIP :BY E
3486 Compressing SIP

St at us :2 00 O
K 4028 Session Timers in SIP
St at us :2 00 O
K
4168 SCTP as a Transport for SIP

4412 Communications Resource Priority for SIP

4566 SDP: Session Description Protocol

GCPs: Gateway Control Protocols A Centralized VOIP Network

Media Gateway Control Protocol (MGCP) and H.248/MEGACO Network Elements
Media Gateway Controller(MGC): also known as Call Agent or Softswitch, it controls a number of dumb terminals and Media Gateways.
The MGC receives signaling information from MG and can instruct it to alert the called party, to send and receive voice data etc.
Media Gateway(MG): acts as a translation unit between disparate telecommunications networks such as PSTN, IP Networks, Mobile
access networks or PBX.
Signaling Gateway (SG): A component responsible for translating signaling messages between IP network and PSTN.
Endpoints: Provide audio, data and video communications with another GCP terminal or a PSTN phone via gateway.

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 Example GCP Call Flow MGCP Documents

Sta nda rd# De s c ri pti on

Vo I P

MGCP MGCP Me d ia Ga t e wa y Co n t ro l Pro t o co l

Ga t e wa y A Ga t e wa y B RFC 3 4 3 5 (MGCP) Ve rsio n 1 . 0

Ca ll
En d p o in t A Ag e n t En d p o in t B
RFC 3 6 6 0 Ba sic MGCP Pa cka g e s
R QN T R QN T

R QN T R es pons e R QN T R es pons e
1
N T F Y f rom A RFC 3 6 6 1 MGCP Re t u rn Co d e Usa g e
Off Ho o k
& Dia llin g CRCX

C R C X R es pons e CRCX
2 R inging RFC 3 0 6 4 MGCP CAS Pa cka g e s
C R C X R es pons e &
Ans w er
MDCX

3 M D C X R es pons e
RFC 3 1 4 9 MGCP Bu sin e ss Ph o n e Pa cka g e s

RTP

RTP RFC 3 9 9 1 MGCP Re d ire ct a n d Re se t Pa cka g e

4
RTCP
On MGCP L o ckst e p St a t e Re p o rt in g
Ho o k N T F Y f rom A RFC 3 9 9 2
Me ch a n ism

5 D LC X D LC X
Me d ia Ga t e wa y Co n t ro l Pro t o co l
D LC X R es pons e D LC X R es pons e RFC 2 8 0 5 Arch it e ct u re a n d Re q u ire me n t s

H.248/Megaco Standards Main difference between Megaco/MGCP

Standard ITU-T File IETF File Description Megaco/H.248 MGCP

A call is represented by teminations A call is represented by endpoints

H.248/ Gateway Control within a call context within connections
Megaco version 1 H.248.1v1 RFC 3525 Protocol Version 1
Call types include any combination Call types include point-to-point
H.248/ of multimedia and conferencing and multipoint
Megaco Version 2 H.248.1v2
Syntax is text binary Syntax is text
H.248/
Megaco Version 3 H.248.1v3
Transport layer is TCP or UDP Transport layer is UDP
Megaco IP Phone
RFC 3054 Media Gateway Defined by the IETF and ITU Defined by Cisco and circulated
Application Profile in IETF

VOIP Media Transport Protocols

Protocol Name Functions Standard#
Real-Time Transport
For delivering audio and video over the Internet. RFC 3550
Protocol (RTP)

Real Time Control

Protocol (RTCP) Provides out-of-band control information for an RTP flow. RFC 3550

Real Time Streaming For use in streaming media systems which allows a client to remotely control
Protocol (RTSP) a streaming media server and allowing time-based access to files on a server. RFC 2326

Reliable User Datagram

Protocol (RUDP) Used as the transport protocol for MGCP based network. RFC 1151

Secure Real-time Transport Provides encryption, message authentication and integrity, and replay protection
Protocol (SRTP) to the RTP data in both unicast and multicast applications. RFC 3711

Stream Control Transmission Provides transport services to ensure reliable, in-sequence transport
of messages with congestion control. RFC 2960
Protocol (SCTP)

An extension to RTP which describes a method of Diffie-Hellman

ZRTP key agreement for SRTP. Draft

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 QoS Technologies VOIP CODECs
Category Technology
Video CODECs
Queuing CODEC Standard
by Applications
Cla ss-Ba se d We ig h t e d Fa ir Qu e u in g (CB-W FQ) Name
Cu st o m Qu e u in g (CQ)

Fa ir Qu e u in g (FQ) Use d p rima rily in o ld e r

H. 2 6 1 I TU-T vid e o co n f e re n cin g a n d
Prio rit y Qu e u in g (PQ) vid e o t e le p h o n y p ro d u ct s.
Prio rit y Qu e u in g - Cla ss-Ba se d We ig h t e d Fa ir Qu e u in g
Ra n d o m e a rly d e t e ct io n o r Ra n d o m e a rly d isca rd (RED) Use d p rima rily f o r
I TU-T vid e o co n f e re n cin g ,
We ig h t e d Fa ir Qu e u in g (W FQ) H. 2 6 3
vid e o t e le p h o n y,
We ig h t e d Ra n d o m Ea rly Dro p / De t e ct (W RED) a n d in t e rn e t vid e o .

Packet
Also kn o wn a s MPEG-4
Classification H. 2 6 4
I TU-T ve rsio n o f
Pa rt 1 0 , o r AVC
MPEG-4 Pa rt 1 0
Typ e o f Se rvice (To S) (f o r Ad va n ce d Vid e o Co d in g ).

Diff Se rv
I n t Se rv
MPEG-4 MPEG Use d f o r in t e rn e t , b ro a d ca st ,
Po licy-b a se d Ro u t in g Pa rt 2 a n d o n st o ra g e me d ia .

Re so u rce Re se rva t io n Pro t o co l (RSVP)

Traffic shaping Also kn o wn a s H. 2 6 4 o r AVC,

andpolicing MPEG-4
Pa rt 1 0
MPEG it is u se d f o r in t e rn e t , b ro a d ca st ,
a n d o n st o ra g e me d ia .
Co mmit t e d Acce ss Ra t e (CAR)
Ge n e ric Tra ff ic Sh a p in g (GTS)

Fragmentation DivX
Ba se d o n Use d f o r in t e rn e t , b ro a d ca st ,
MPEG-4 Pa rt 2 a n d o n st o ra g e me d ia .
Mu lt i-Cla ss Mu lt ilin k Po in t -t o -Po in t Pro t o co l (MCML PPP)
Fra me Re la y Fo ru m 1 2 (FRF. 1 2 )
Ma ximu m Tra n smissio n Un it (MTU) I t ca n d o a n yt h in g f ro m
WMV Windows Micro so f t lo w re so lu t io n vid e o f o r
Other Media Video d ia l u p in t e rn e t u se rs t o HDTV.
Co mp re sse d Re a l Time Tra n sp o rt Pro t o co l (cRTP)

X2 6 4 Ba se d o n A GPL -lice n se d imp le me n t a t io n

H. 2 6 4 ; GPL o f H. 2 6 4 e n co d in g st a n d a rd .
Audio CODECs
Codec Standard Modulation Bit rate Sampling Frame size Compression Mean Opinion
Type by method (kb/s) rate (kHz) (ms) delay Score (MOS) Notes

U-la w (US, Ja p a n ) a n d A-la w

G. 7 11 I TU-T PCM 64 8 0.125 0.75 4.1 (Eu ro p e ) co mp a n d in g .
G. 7 2 1 I TU-T ADPCM 32 8 Sa mp lin g Re p la ce d b y G. 7 2 6 .
Su b b a n d -co d e c t h a t d ivid e s 1 6 kHz
G. 7 2 2 I TU-T ADPCM 64 16 Sa mp lin g b a n d in t o t wo su b b a n d s, e a ch co d e d
u sin g ADPCM.
G. 7 2 2 . 1 I TU-T Tra n sf o rm-b a se d 24/32 16 20 40
6.60, 8.85, 12.65,
14.25,15.85, 18.25,, Sa mp lin g AMR-W B is st a n d a rd ize d f o r u sa g e
G. 7 2 2 . 2 I TU-T AMR-W B 19.85, 23.05 16 in n e t wo rks su ch a s UMTS.
and 23.85

Su p e rce d e d b y G. 7 2 6 ; Th is is a
G. 7 2 3 I TU-T DPCM 24/40 8 Sa mp lin g 30 co mp le t e ly d iff e re n t co d e c t h a n
G. 7 2 3 . 1

MPC-ML Q o r
G. 7 2 3 . 1 I TU-T 5.6/6.3 8 30 30 3 . 7 -3 . 9 Pa rt o f H. 3 2 4 vid e o co n f e re n cin g .
ACEL P
G. 7 2 6 I TU-T ADPCM 16/24/32/40 8 0.125 1 3.9 Re p la ce s G. 7 2 1 a n d G. 7 2 3 .
G. 7 2 7 I TU-T ADPCM 5 -, 4 -, 3 - a n d 2 8 Sa mp lin g Re la t e d t o G. 7 2 6 .
G. 7 2 8 I TU-T L DCEL P 16 8 0.625 0.625 3.6
G. 7 2 9 CS-ACEL P 8 2.15-24.6 (NB) 10 15 3.9 VOI P Ap p lica t io n s.

Spe e x 3 0 ( NB )
Fre e wa re CEL P 8, 16, 32 4-44.2 (WB) 3 0– 3 4 VOI P
34 ( WB )
iLBC I ETF RFC L PC 8, 16 13.3 30 30 4.0
(Internet 3951 VOI P
Low Bitrate Fre e wa re L PC 8, 16 15.2 20 20 4.0
Vocoder)
I ETF RFC Uncompressed
L1 6 128 Va ria b le Sa mp lin g
3551 audio data samples

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 VOIP Glossar y
ACELP--Algebraic Code Excited Linear Prediction Modulation--To carring information on a signal by varying one or more of the signal's basic characteristics --
ADPCM--Adaptive Differential Pulse Code Modulation frequency, amplitude and phase.
AMR-WB--Adaptive Multi Rate – WideBand MoIP--Modem over IP.
ATA--Analog Telephone Adaptor connects the conventional telephone to the Internet, converts the analog voice MOS--Mean Opinion Score, a numerical indication of the perceived quality of received media after
signals into IP packets, delivers dial tone and manages the call setup. compression and/or transmission.
Broadband--High speed Internet connection, such as cable TV, DSL or dedicated telecom lines(T1/E1). MPEG-4 Part 10--Also known as H.264 or AVC, a video codec used for internet, broadcast, and on storage
C7--Common Channel Signaling 7. media.
Cable modem--A device used to connect a computer to the high speed coaxial cable run by cable TV companies MPEG-4 Part 2--Used for internet, broadcast, and on storage media.
to provide access to the Internet. MP-MLQ--Multi-Pulse, Multi-Level Quantization.
CALEA--Communications Assistance for Law Enforcement Act. MTU--Maximum Transmission Unit.
Call agent--The intelligent and controlling entity in an MCGP based IP telephony network. NCS--Network-Based Call Signaling.
Call flow--The setup and tear down process and steps for a call to start till finish. Net Phone --A net phone uses the Voice over IP technology to make voice calls.
CB-WFQ--Class-Based Weighted Fair Queuing. Network convergence--The integration of all traffic types - voice, data and video solutions - onto a single IP
CDR--Call Detail Record. network.
CID--Caller Identification (ID). NGN--Next Generation Network.
Circuit switched network--The traditional telephone network used for making phone calls since 1878. OSP--Open Settlement Protocol.
Codec--Compressor-Decompressor or enCOder/DECoder process. Packet loss--The loss of data packets during transmission over a computer network.
Committed Access Rate (CAR)--A QoS feature. Packet switched network--Networks that break messages into small packets, and route them across different
Compression--The squeezing of data in a format that takes less space to store or less bandwidth to transmit. channels to their destination where they are reassembled in their proper sequence.
Compression delay--The delay caused by the compression of data. PBX--Private Branch Exchange is an in-house telephone switching system.
Congestion--The situation in which the traffic present on the network exceeds available network PCM--Pulse Code Modulation.
bandwidth/capacity. Peer-to-Peer (P2P)--A form of computing where two or more than two users can communicate directly without a
CoS--Class of Service. central control point.
CPE--Customer Premises Equipment. Policy-based Routing--A technique used to make routing decisions based on policies set by the network
cRTP--Compressed Real Time Transport Protocol. administrator.
CS-ACLEP--Conjugate-Structure Algebraic-Code-Excited Linear-Prediction. POTS--Plain Old Telephone Service.
Custom Queuing-- A queuing method that allows a customer to reserve a percentage of bandwidth for specified PQ-CBWFQ--Priority Queuing - Class-Based Weighted Fair Queuing.
protocols. PRI--Primary Rate Interface, a type of ISDN interface.
Data compression-- The process to compress large data files into small files so that they use less bandwidth Priority Queuing (PQ)--A queuing technique to give mission-critical traffic higher priority that less critical
during transmission and less disk space when stored. traffic.
Decompression--Process by which the full data content of a compressed file is restored. Processor drain--A drop in the quality of VoIP phone service when a user opens several applications on his
DiffServ-- An architecture for implementing scalable service differentiation in the Internet for QoS. computer simultaneously.
DivX-- A video codec used for internet, broadcast, and on storage media. Propogation delay--The time required for a signal to travel from one point to another.
DPCM--Differential Pulse Code Modulation. Proxy server-- Performs routing of a session invitations according to invitee's current location, authentication,
DSL modem-- A device used to connect computers to the DSL line provided by a DSL operator to gain access to accounting, etc.
the Internet. PSTN--Public Switched Telephone Network, refers to the telephone system that transmits analog voice data.
DTMF--Dual-Tone Multi Frequency. Q.931--ISDN connection control protocol.
Dynamic Jitter Buffer-- Collects voice packets, stores them, and shifts them to the voice processor in evenly QoS--Quality of Service.
spaced intervals to reduce any distortion in the sound. QSIG--Signaling standard for PBX.
E&M (Ear and Mouth)--A type of supervisory line signaling. RADIUS--Remote Authenticaion Dial-In User Service.
E911--Enhanced 911; used for providing emergency service on cellular and Internet voice calls. Random Early Detection (RED)--An active queue management algorithm. It is also a congestion avoidance
Emergency 911 calls--An emergency telephone number that handles all calls related to police, fire or medical algorithm.
emergencies in North America. RAS--Registration, Admission, Status (RAS), a management protocol between terminals and Gatekeepers in
Fair Queuing--A scheduling scheme to allow several data flows to fairly share the link capacity. the H.323 network.
FoIP--Fax over Internet Protocol. Redirect server--Receives a request and sends back a reply containing a list of the current location of a
Frame Relay Forum 12 (FRF.12)--A Frame Relay specification of fragmenting Frame Relay frames into smaller particular user.
frames. Registrar server-- Accepts REGISTER requests and places the information it receives in those requests into
G.711--ITU-T specification of audio CODEC. G.721--ITU-T specification of audio CODEC. the location service for the domain in handles.
G.722--ITU-T specification of audio CODEC. G.722.1--ITU-T specification of audio CODEC. RSVP-- Resource Reservation Protocol.
G.722.2--ITU-T specification of audio CODEC. G.723--ITU-T specification of audio CODEC. RTCP-- Real Time Control Protocol.
G.723.1--ITU-T specification of audio CODEC. G.726--ITU-T specification of audio CODEC. RTP-- Real Time Transport Protocol.
G.727--ITU-T specification of audio CODEC. G.728--ITU-T specification of audio CODEC. RTSP--Real Time Streaming Protocol
G.729--ITU-T specification of audio CODEC. RUDP--Reliable User Datagram Protocol.
Gatekeeper--A device that translates network addresses and aliases to make connections via the H.323 protocol Sampling--A methodology used to measure the value of an analog signal at regular intervals, and encoding it
on a packet-switched network. into a digital format for phone services.
Gateway--A device that acts as an interface between two or more networks to connect dissimilar communications Sampling rate--The number of samples per second taken from a continuous (analog) signal to make a
systems. discrete(digital) signal.
Generic Traffic Shaping (GTS)--A mechanism to control the traffic flow on a particular interface. SAP-- Session Announcement Protocol.
H.225.0--A protocol for call signaling, the media (audio and video), the stream packetization, media stream SCCP--Skinny Client Control Protocol.
synchronization and control message formats. SCTP--Stream Control Transmission Protocol.
H.235--For security in H.323 network. SDP--Session Description Protocol.
H.239--For dual stream use in videoconferencing. Servie Level Agreement (SLA)--A contract between a network service provider and a customer that specifies
H.245--A control protocol for multimedia communication. what services and quality the service provider will furnish.
H.246--ITU-T specification for H.323/PSTN Interworking. Service provider-- A business entity that provides a communication, storage or processing service for a fee.
H.248--ITU-T standard for a centralized VOIP network. (Same as Megaco defined by IETF.) SGCP-- Simple Gateway Control Protocol.
H.261--Used primarily in older videoconferencing and video telephony products. Signaling gateway--A network component responsible for translating signaling messages between one
H.263--Used primarily for videoconferencing, video telephony, and internet video. medium (usually IP) and another (PSTN).
H.264--Also known as MPEG-4 Part 10, or AVC (for Advanced Video Coding). SIGTRAN-- A family of protocols that provides reliable datagram service and user layer adaptations for SS7
H.323--An umbrella recommendation from the ITU-T that defines the protocols to provide audio-visual and ISDN communications protocols.
communication sessions on any packet network. SIP--Session Initiation Protocol, an IP telephony signaling protocol.
H.350--Directory Services Architecture for Multimedia Conferencing. SIP phone-- A telephone that uses the SIP (Session Initiation Protocol) standard to make a voice call over the
H.360--An architecture for end-to-end QoS control and signaling. Internet.
H.450--For supplementary services such as call waiting, call forwarding, etc. Skinny-- Skinny Client Control Protocol.
H.460.x--Supplements in H.323. Skype--A peer-to-peer Internet telephony company that leading the way voice calls are made by using VoIP
H.501--Protocol for mobility management and intra/inter-domain communication in multimedia systems. technology.
H.510--Mobility for H.323 multimedia systems. Soft switch-- A software application that is used to keep track of, monitor or regulate connections at the
Hairpin--To send a call back in the direction that it came from. junction point between circuit and packet networks.
Hop off--Point at which a call transitions from H.323 to non-H.323, typically at a gateway. Softphone-- A software application that is installed in the user’s PC enables voice calls over the Internet.
iLBC--Internet Low Bitrate Vocoder. Softphone client-- The software installed in the user’s computer to make calls over the Internet.
Instant Messenging (IM)--A software that allows users to exchange messages in real time. For example, MSN Speex--A free software speech codec.
Messenger, Yahoo! Messenger, etc. SRTP--Secure Real-time Transport Protocol.
Internet telephony--Technologies and services of using the Internet for voice and multimedia communications. SS7--Signaling System number 7.
IntServ--An architecture which specifies the elements to guarantee quality of service (QoS) on networks. T.120-- ITU-T specifcation for Real-time Conferencing.
IP--Inernet Protocol. T.38-- ITU-T specification for Facsimile over IP.
IP Centrex--Using IP-based network to provide centrex services such as call hold, call transfer, last number TAPI--Telephony API.
look-up and redial, call forward, three-way calling. ToS--Type of Service.
IP fragmentation--IP datagrams to be fragmented into pieces small enough to pass over a link with a smaller Traffic shaping--To control network traffic in order to optimize or guarantee performance, low latency, and/or
MTU than the original datagram size. bandwidth
IP PBX--IP Private Branch Exchange. A telephone, data and video switching system, usually located on customer Unified Messaging (UM)-- The integration of different streams of messages (e-mail, Fax, voice, video, etc.)
premises and belonging to the user. into a single in-box, accessible from a variety of different devices.
IP phone--A device that converts voice into digital packets and vice versa to make phone calls over Internet User Agents-- A software program installed in a user’s terminal or an IP phone to initiate and terminate phone
possible. calls.
IP telephony--Technologies and services for the two-way transmission of voice over IP network. V.150-- ITU-T specification for Modem over IP.
IPDC--IP Device Control (protocol). Voice chat-- An application that enables two or more individuals to carry on a verbal conversation (audio
IPDR--Internet Protocol Detail Record (protocol). conference) over the Internet.
ISDN--Integrated Services Digital Network. Voice over IP (VOIP)--The technology that is used to transmit voice over the Internet.
ITSP--Internet Telephony Service Provider. Voicemail-- A telephone messaging system that digitizes the analog voice signals and stores them on disk or
Jitter--A momentary fluctuation in the transmission signal. flash memory in a central computer.
Lag--The extra time taken by a packet of data to travel from the source computer to the destination computer and VOIP Gateway-- A device provides the conversion interface between the PSTN and an IP network for voice and
back again. fax calls.
Latency--The time that elapses between the initiation of a request for data and the start of the actual data VOIP PBX-- Voice over Internet Protocol Private Branch eXchange.
transfer. VOIP Phone-- A device that uses the IP network to route voice calls by converting the voice data into IP packets
LDCELP--Low-Delay Code Excited Linear Prediction. and vice versa.
LPC--Linear-Predictive Codec. VOIP services-- Services that use the IP network to move voice data.
LPCP--Lightweight Phone Control Protocol. Web phone-- A device that allows users to make voice calls over the Internet.
MCML PPP--Multi-Class Multilink Point-to-Point Protocol. WFQ-- Weighted Fair Queuing, a packet scheduling technique allowing guaranteed bandwidth services.
Media gateway (MG)--A translation unit between disparate telecommunications networks. WiFi phone-- A device that enables users to make phone calls from WiFi network environments.
Media gateway controller (MGC)--A system used in MGCP/H.248/Megaco VoIP telephony architectures to WMV-- Windows Media Video.
control a number of Media Gateways. WRED--Weighted Random Early Drop/Detect.
Megaco--A IETF VOIP signaling protocol, same as H.248 of ITU-T. X264-- A GPL-licensed implementation of H.264 encoding standard.
MGCP--Media Gateway Control Protocol. ZRTP-- An extension to RTP which describes a method of Diffie-Hellman key agreement for SRTP.

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